GN: Update xmpp and p2p to match Chromium build

webrtc/p2p/BUILD.gn

 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../build/webrtc.gni")
 
 group("p2p") {
   deps = [
     ":rtc_p2p",
   ]
 }
 
 config("rtc_p2p_inherited_config") {
   defines = [ "FEATURE_ENABLE_VOICEMAIL" ]
 }
 
 source_set("rtc_p2p") {
   sources = [
     "base/asyncstuntcpsocket.cc",
     "base/asyncstuntcpsocket.h",
     "base/basicpacketsocketfactory.cc",
     "base/basicpacketsocketfactory.h",
     "base/candidate.h",
     "base/common.h",
+    "base/dtlstransport.h",
     "base/dtlstransportchannel.cc",
     "base/dtlstransportchannel.h",
     "base/p2pconstants.cc",
     "base/p2pconstants.h",
     "base/p2ptransport.cc",
     "base/p2ptransport.h",
     "base/p2ptransportchannel.cc",
     "base/p2ptransportchannel.h",
     "base/packetsocketfactory.h",
     "base/port.cc",
     "base/port.h",
     "base/portallocator.cc",
     "base/portallocator.h",
     "base/portinterface.h",
     "base/pseudotcp.cc",
     "base/pseudotcp.h",
     "base/relayport.cc",
     "base/relayport.h",
-    "base/relayserver.cc",
-    "base/relayserver.h",
+    "base/session.cc",
+    "base/session.h",
     "base/sessiondescription.cc",
     "base/sessiondescription.h",
     "base/sessionid.h",
     "base/stun.cc",
     "base/stun.h",
     "base/stunport.cc",
     "base/stunport.h",
     "base/stunrequest.cc",
     "base/stunrequest.h",
-    "base/stunserver.cc",
-    "base/stunserver.h",
     "base/tcpport.cc",
     "base/tcpport.h",
     "base/transport.cc",
     "base/transport.h",
     "base/transportchannel.cc",
     "base/transportchannel.h",
     "base/transportchannelimpl.h",
     "base/transportcontroller.cc",
     "base/transportcontroller.h",
     "base/transportdescription.cc",
     "base/transportdescription.h",
     "base/transportdescriptionfactory.cc",
     "base/transportdescriptionfactory.h",
     "base/transportinfo.h",
     "base/turnport.cc",
     "base/turnport.h",
-    "base/turnserver.cc",
-    "base/turnserver.h",
     "base/udpport.h",
     "client/basicportallocator.cc",
     "client/basicportallocator.h",
     "client/httpportallocator.cc",
     "client/httpportallocator.h",
     "client/socketmonitor.cc",
     "client/socketmonitor.h",
   ]
 
   defines = [ "FEATURE_ENABLE_SSL" ]
 
   deps = [
     "../base:rtc_base",
     "../libjingle/xmllite",
   ]
 
   if (rtc_build_expat) {
     deps += [ "//third_party/expat" ]
     public_deps = [
       "//third_party/expat",
     ]
   }
 
   configs += [ "..:common_config" ]
   public_configs = [
     "..:common_inherited_config",
     ":rtc_p2p_inherited_config",
   ]
 
-  if (!build_with_chromium) {
+  if (build_with_chromium) {
+    if (is_nacl) {
+      deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
+    }
+  } else {
+    sources += [
+      "base/relayserver.cc",
+      "base/relayserver.h",
+      "base/stunserver.cc",
+      "base/stunserver.h",
+      "base/turnserver.cc",
+      "base/turnserver.h",
+    ]
     defines += [
       "FEATURE_ENABLE_VOICEMAIL",
       "FEATURE_ENABLE_PSTN",
     ]
 
     if (is_clang) {
       # Suppress warnings from Chrome's Clang plugins.
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       configs -= [ "//build/config/clang:find_bad_constructs" ]
     }
@@ skipping 28 matching lines @@
     # Suppress warnings from Chrome's Clang plugins.
     # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     "..:webrtc_common",
     "../base:rtc_base",
   ]
 }